Apparatus and method for encoding a multi channel audio signal

ABSTRACT

An encoding apparatus comprises a frame processor ( 105 ) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone ( 101 ) and a second audio signal from a second microphone ( 103 ). An ITD processor  107  then determines an inter time difference between the first audio signal and the second audio signal and a set of delays ( 109, 111 ) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner ( 113 ) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder ( 115 ) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a Continuation of, and claims priority to, U.S.patent application Ser. No. 13/920,549, filed on Jun. 18, 2013, which isa Continuation of, and claims priority to, U.S. patent application Ser.No. 12/679,121, filed on Dec. 14, 2010, now U.S. Pat. No. 8,577,045,which is a 371 of, and claims priority to, International Application No.PCT/US08/75703, filed on Dec. 14, 2010, which claims priority to UnitedKingdom Patent Application No. 0718682.8, filed on Sep. 25, 2007, thedisclosures of which are incorporated by reference herein in theirentirety.

FIELD OF THE INVENTION

The invention relates to an apparatus and method for encoding a multichannel audio signal and in particular, but not exclusively, to down-mixa stereo speech signal to a mono signal for encoding with a monoencoder, such as a Code Excited Linear Prediction encoder.

BACKGROUND OF THE INVENTION

Efficient encoding of audio signals is critical for an increasing numberof applications and systems. For example, mobile communications useefficient speech encoders to reduce the amount of data that needs to betransmitted over the air interface.

For example, the International Telecommunication Union (ITU) isstandardizing a speech encoder known as the Embedded Variable Bit RateCodec (EV-VBR) which can encode a speech signal at high quality withdata rates ranging from 8 to 64 kbps. This encoder, as well as manyother efficient speech encoders, uses Code Excited Linear Prediction(CELP) techniques to achieve the high compression ratio of the encodingprocess at the lower bit rates of operation.

In some applications, more than one audio signal may be captured and inparticular a stereo signal may be recorded in audio systems using twomicrophones. For example, stereo recording may typically be used inaudio and video conferencing as well as broadcasting applications.

In many multi channel encoding systems, and in particular in many multichannel speech encoding systems, the low level encoding is based onencoding of a single channel. In such systems, the multi channel signalmay be converted to a mono signal for the lower layers of the coder toencode. The generation of this mono signal is referred to asdown-mixing. Such down-mixing may be associated with parameters thatdescribe aspects of the stereo signal relative to the mono signal.Specifically, the down mixing may generate inter-channel time difference(ITD) information which characterises the timing difference between theleft and right channels. For example, if the two microphones are locatedat a distance from each other, the signal from a speaker located closerto one microphone than the other will reach the latter microphone with adelay relative to the first one. This ITD may be determined and may inthe decoder be used to recreate the stereo signal from the mono signal.The ITD may significantly improve the quality of the recreated stereoperspective since ITD has been found to be the dominant perceptualinfluence on stereo location for frequencies below approximately 1 kHz.Thus, it is critical that ITD is also estimated.

Conventionally, the mono signal is generated by summing the stereosignals together. The mono signal is then encoded and transmitted to thedecoder together with the ITD.

For example, the European Telecommunication Standards Institute has intheir Technical Specification ETSI TS126290 “Extended AdaptiveMulti-Rate-Wideband (AMR-WB+) Codec; Transcoding Functions” defined astereo signal down-mixing where the mono signal is simply determined asthe average of the left and right channels as follows.

x _(ML)(n)=0.5(x _(LL)(n)+x _(RL)(n))

where x_(ML)(n) represents the nth sample of the mono signal, x_(LL)(n)represents the nth sample of the left channel signal and x_(RL)(n)represents the nth sample of the right channel signal.

Another example of a downmix is provided in H. Purnhagen, “LowComplexity Parametric Stereo Coding in MPEG-4”, Proceedings 7^(th)International Conference on Digital Audio Effects (DAFx'04), Naples,Italy, Oct. 5-8, 2004, pp 163-168. In this document, a down-mixingmethod is described which obtains an output mono signal as a weightedsum of the incoming channels on a band-by-band frequency basis usinginformation obtained about the inter-channel intensity difference (IID).Specifically:

M[k,i]=g _(l) L[k,i]+g _(r) R[k,i]

where M[k,i] represents the ith sample of the kth frequency bin of monosignal, L[k,i] represents the ith sample of the kth frequency bin of theleft channel signal and R[k,i] represents the ith sample of the kthfrequency bin of the right channel signal, g_(l) is the left channelweight and g_(r) is the right channel weight.

A characteristic of such approaches is that they either result in monosignals having a high reverberation time or else have high complexityand/or delay. For example, the AMR-WB+ method of down-mixing provides anoutput whose reverberation time is approximately that of the room plusthe flight time between the two microphones. The downmix provided inPurnhagen is of high complexity and imposes a delay due to the frequencyanalysis and reconstruction.

However, many mono encoders provide the best results for signals withlow reverberation times. For example, low bit rate CELP speech coders,and other encoders which employ pulse-based excitation to representspeech and audio signals, perform best when presented with signalshaving short reverberation times. Accordingly, the performance of theencoder and the quality of the resulting encoded signal tend to besuboptimal.

Hence, an improved system would be advantageous and in particular asystem allowing increased flexibility, facilitated implementation,improved encoding quality, improved encoding efficiency, reduced delayand/or improved performance would be advantageous.

SUMMARY OF THE INVENTION

Accordingly, the Invention seeks to preferably mitigate, alleviate oreliminate one or more of the above mentioned disadvantages singly or inany combination.

According to an aspect of the invention there is provided an apparatusfor encoding a multi channel audio signal, the apparatus comprising: areceiver for receiving the multi channel audio signal comprising atleast a first audio signal from a first microphone and a second audiosignal from a second microphone; a time difference unit for determiningan inter time difference between the first audio signal and the secondaudio signal; a delay unit for generating a compensated multi channelaudio signal from the multi channel audio signal by delaying at leastone of the first audio signal and the second audio signal in response tothe inter time difference signal; a mono unit for generating a monosignal by combining channels of the compensated multi channel audiosignal; and a mono signal encoder for encoding the mono signal.

The invention may provide improved encoding of a multi channel audiosignal. In particular, an improved quality for a given data rate may beachieved in many embodiments. The invention may provide improved monoencoding of a mono down mix signal from a stereo signal by reducingreverberation times of the mono down mix signal. The delay unit maydelay either the first audio signal or the second audio signal dependingon which microphone is closest to the (main) audio source. The intertime difference may be an indication of a time difference betweencorresponding audio components of the first and second audio signalsoriginating from the same audio source. The unit for generating the monosignal may be arranged to sum the two channels of the combined multichannel audio signal which correspond to the first and second audiosignals. In some embodiments, the summation may be a weighted summation.

According to an optional feature of the invention, the time differenceunit is arranged to determine cross correlations between the first audiosignal and the second audio signal for a plurality of time offsets, andto determine the inter time difference in response to the crosscorrelations.

The feature may allow an improved determination of the inter timedifference. The feature may improve the quality of the encoded audiosignal and/or may facilitate implementation and/or reduce complexity. Inparticular, the feature may allow improved stereo perception of a stereosignal rendered from the mono signal and the inter time difference. Thecross correlations may indicate a probability of the inter timedifference being equal to the time offset of the individual crosscorrelations.

According to another aspect of the invention there is provided a methodof encoding a multi channel audio signal, the method comprising:receiving the multi channel audio signal comprising at least a firstaudio signal from a first microphone and a second audio signal from asecond microphone; determining an inter time difference between thefirst audio signal and the second audio signal; generating a compensatedmulti channel audio signal from the multi channel audio signal bydelaying at least one of the first audio signal and the second audiosignal in response to the inter time difference signal; generating amono signal by combining channels of the compensated multi channel audiosignal; and encoding the mono signal in a mono signal encoder.

These and other aspects, features and advantages of the invention willbe apparent from and elucidated with reference to the embodiment(s)described hereinafter.

BRIEF DESCRIPTION OF THE DRAWINGS

Embodiments of the invention will be described, by way of example only,with reference to the drawings, in which

FIG. 1 illustrates an example of an apparatus for encoding a multichannel audio signal in accordance with some embodiments of theinvention;

FIG. 2 illustrates an example of a processing unit for estimating aninter time difference in accordance with some embodiments of theinvention;

FIG. 3 illustrates an example of a whitening processor in accordancewith some embodiments of the invention;

FIG. 4 illustrates an example of a state update for a trellis statemachine in accordance with some embodiments of the invention; and

FIG. 5 illustrates an example of a method for encoding a multi channelaudio signal in accordance with some embodiments of the invention.

DETAILED DESCRIPTION OF SOME EMBODIMENTS OF THE INVENTION

The following description focuses on embodiments of the inventionapplicable to encoding of a multi channel audio signal using a monoencoder and in particular to encoding of a stereo speech signal using amono CELP encoder.

FIG. 1 illustrates an apparatus for encoding a multi channel audiosignal in accordance with some embodiments of the invention. In thespecific example, a stereo speech signal is down-mixed to a mono signaland encoded using a mono encoder.

The apparatus comprises two microphones 101, 103 which capture audiosignals from the audio environment in which the two microphones arelocated. In the example, the two microphones are used to record speechsignals in a room and are located with an internal distance of up to 3meters. In the specific application, the microphones 101, 103 may forexample be recording speech signals from a plurality of people in theroom and the use of two microphones may provide better audio coverage ofthe room.

The microphones 101, 103 are coupled to a frame processor 105 whichreceives the first and second signals from the first and secondmicrophones 101, 103 respectively. The frame processor divides thesignals into sequential frames. In the specific example, the samplefrequency is 16 ksamples/sec and the duration of a frame is 20 msecresulting in each frame comprising 320 samples. It should be noted thatthe frame processing need not result in an additional delay to thespeech path since this frame may be the same frame as that used forspeech encoding or the frame processing may e.g. be performed on oldspeech samples.

The frame processor 105 is coupled to an ITD processor 107 which isarranged to determine an inter time difference between the first audiosignal and the second audio signal. The inter time difference is anindication of the delay of the signal in one channel relative to thesignal in the other. In the example, the inter time difference may bepositive or negative depending on which of the channels is delayedrelative to the other. The delay will typically occur due to thedifference in the delays between the dominant speech source (i.e. thespeaker currently speaking) and the microphones 101, 103.

The ITD processor 107 is furthermore coupled to two delays 109, 111. Thefirst delay 109 is arranged to introduce a delay to the first audiochannel and the second delay 109 is arranged to introduce a delay to thesecond audio channel. The amount of the delay which is introduceddepends on the estimated inter time difference. Furthermore, in thespecific example only one of the delays is used at any given time. Thus,depending on the sign of the estimated inter time difference, the delayis either introduced to the first or the second audio signal. The amountof delay is specifically set to be as close to the estimated inter timedifference as possible. As a consequence, the audio signals at theoutput of the delays 109,111 are closely time aligned and willspecifically have an inter time difference which typically will be closeto zero.

The delays 109, 111 are coupled to a combiner 113 which generates a monosignal by combining the channels of the compensated multi channel audiosignal and specifically by combining the two output signals from thedelays 109, 111. In the example, the combiner 113 is a simple summationunit which adds the two signals together. Furthermore, the signals arescaled by a factor of 0.5 in order to maintain the amplitude of the monosignal similar to the amplitude of the individual signals prior to thecombination.

Thus, the output of the combiner 113 is a mono signal which is adown-mix of the two captured signals. Furthermore, due to the delay andthe reduction of the inter time difference, the generated mono signalhas significantly reduced reverberation.

The combiner 113 is coupled to a mono encoder 115 which performs a monoencoding of the mono signal to generate encoded data. In the specificexample, the mono encoder is a Code Excited Linear Prediction (CELP)encoder in accordance with the Embedded Variable Bit Rate Codec (EV-VBR)to be standardised by the International Telecommunication Union (ITU).

CELP coders are known to provide extremely efficient encoding andspecifically to provide good speech quality even for low data rates.However, CELP coders tend not to perform as well for signals with highreverberation times and have therefore not been suitable for encoding ofconventionally generated mono down mixes. However, due to the delaycompensation and resulting reduced reverberation, CELP mono encoders maybe used in the apparatus of FIG. 1 to provide a very efficient encodingof a speech down mix mono signal. It will be appreciated that theseadvantages are particularly appropriate for CELP mono encoders but arenot limited thereto and may apply to many other encoders.

The mono encoder 115 is coupled to an output multiplexer 117 which isfurthermore coupled to the ITD processor 107. In the example, the outputmultiplexer 117 multiplexes the encoding data from the mono encoder 115and data representing the inter time difference from the ITD processor107 into a single output bitstream. The inclusion of the inter timedifference in the bitstream may assist the decoder in recreating astereo signal from a mono signal decoded from the encoding data.

Thus, the described system provides improved performance and may inparticular provide an improved audio quality for a given data rate. Inparticular, the improved use of a mono encoder such as a CELP encodermay result in significantly improved quality. Furthermore, the describedfunctionality is simple to implement and has relatively low resourcerequirements.

In the following, the inter time difference estimation performed by theITD processor 107 will be described with reference to FIG. 2.

The algorithm used by the ITD processor 107 determines an estimate ofthe inter time difference by combining successive observations ofcross-correlations between the first and second audio signals fordifferent possible time offsets between the channels. The correlationsare performed in a decimated LPC residual domain in order to providemore well defined correlations, facilitate implementation and reduce thecomputational demands. In the example, the cross-correlations areprocessed to derive a probability associated with each potential delaybetween −12 ms and +12 ms (±˜4 meters) and the probabilities are thenaccumulated using a modified Viterbi-like algorithm. The result is anestimate of the inter time difference with in-built hysteresis.

The ITD processor 107 comprises a decimation processor 201 whichreceives the frames of samples for the two channels from the frameprocessor 105. The decimation processor 201 first performs a low passfiltering followed by a decimation. In the specific example, the lowpass filter has a bandwidth of around 2 kHz and a decimation factor offour is used for a 16 ksamples/sec signal resulting in a decimatedsample frequency of 4 ksamples/sec. The effect of the filtering anddecimation is partly to reduce the number of samples processed therebyreducing the computational demand. However, in addition, the approachallows the inter time difference estimation to be focussed on lowerfrequencies where the perceptual significance of the inter timedifference is most significant. Thus, the filtering and decimation notonly reduces the computational burden but also provides the synergisticeffect of ensuring that the inter time difference estimate is relevantto the most sensitive frequencies.

The decimation processor 201 is coupled to a whitening processor 203which is arranged to apply a spectral whitening algorithm to the firstand second audio signals prior to the correlation. The spectralwhitening leads to the time domain signals of the two signals moreclosely resembling a set of impulses, in the case of voiced or tonalspeech, thereby allowing the subsequent correlation to result in morewell defined cross correlation values and specifically to result innarrower correlation peaks (the frequency response of an impulsecorresponds to a flat or white spectrum and conversely the time domainrepresentation of a white spectrum is an impulse).

In the specific example, the spectral whitening comprises computinglinear predictive coefficients for the first and second audio signalsand to filter the first and second audio signals in response to thelinear predictive coefficients.

Elements of the whitening processor 203 are shown in FIG. 3.Specifically, the signals from the decimation processor 201 are fed toLPC processors 301, 303 which determine Linear Predictive Coefficients(LPCs) for linear predictive filters for the two signals. It will beappreciated that different algorithms for determining LPCs will be knownto the skilled person and that any suitable algorithm may be usedwithout detracting from the invention.

In the example, the two audio signals are fed to two filters 305, 307which are coupled to the LPC processors 301, 303. The two filters aredetermined such that they are the inverse filters of the linearpredictive filters determined by the LPC processors 301, 303.Specifically, the LPC processors 301, 303 determine the coefficients forthe inverse filters of the linear predictive filters and thecoefficients of the two filters are set to these values.

The output of the two inverse filters 305, 307 resemble sets of impulsetrains in the case of voiced speech and thereby allow a significantlymore accurate cross-correlation to be performed than would be possiblein the speech domain.

The whitening processor 203 is coupled to a correlator 205 which isarranged to determine cross correlations between the output signals ofthe two filters 305, 307 for a plurality of time offsets.

Specifically, the correlator can determine the values:

$c^{t} = {\sum\limits_{N}\; {x_{n} \cdot y_{n - t}}}$

where t is the time offset, x and y are samples of the two signals and Nrepresents the samples in the specific frame.

The correlation is performed for a set of possible time offsets. In thespecific example, the correlation is performed for a total of 97 timeoffsets corresponding to a maximum time offset of ±12 msec. However, itwill be appreciated that other sets of time offsets may be used in otherembodiments.

Thus, the correlator generates 97 cross-correlation values with eachcross-correlation corresponding to a specific time offset between thetwo channels and thus to a possible inter time difference. The value ofthe cross-correlation corresponds to an indication of how closely thetwo signals match for the specific time offset. Thus, for a high crosscorrelation value, the signals match closely and there is accordingly ahigh probability that the time offset is an accurate inter timedifference estimate. Conversely, for a low cross correlation value, thesignals do not match closely and there is accordingly a low probabilitythat the time offset is an accurate inter time difference estimate.Thus, for each frame the correlator 205 generates 97 cross correlationvalues with each value being an indication of the probability that thecorresponding time offset is the correct inter time difference.

In the example, the correlator 205 is arranged to perform windowing onthe first and second audio signals prior to the cross correlation.Specifically, each frame sample block of the two signals is windowedwith a 20 ms window comprising a rectangular central section of 14 msand two Hann portions of 3 ms at each end. This windowing may improveaccuracy and reduce the impact of border effects at the edge of thecorrelation window.

Also, in the example, the cross correlation is normalised. Thenormalisation is specifically to ensure that the maximumcross-correlation value that can be achieved (i.e. when the two signalsare identical) has unity value. The normalisation provides forcross-correlation values which are relatively independent of the signallevels of the input signals and the correlation time offsets testedthereby providing a more accurate probability indication. In particular,it allows improved comparison and processing for a sequence of frames.

In a simple embodiment, the output of the correlator 205 may directly beevaluated and the inter time difference for the current frame may be setto the value which has the highest probability as indicated by the crosscorrelation value. However, such a method would tend to provide a lessreliable output as the speech signal fluctuates from voiced to unvoicedto silence and in the described example, the correlator is fed to astate processor 207 which processes correlation values for a pluralityof states to provide a more accurate inter time difference estimate.

In the example the correlation values are used as update steps to aViterbi algorithm metric accumulator implemented in the state processor207.

Thus, the state processor 207 specifically implements a metricaccumulator which has a number of states corresponding to the timeoffsets. Each state thus represents a time offset and has an associatedaccumulated metric value.

Accordingly, a Viterbi based trellis state machine in the form of themetric accumulator stores a metric value for each of the time offsetsfor which a correlation value has been calculated (i.e. 97 states/timeoffstets in the specific example). Each state/time offset isspecifically associated with a probability metric which is indicative ofthe probability that the inter time difference corresponds to the timeoffset of that state.

The probability metrics for all time offsets are recalculated in everyframe to take into account the correlation values which have beendetermined for the current frame. Specifically, path metrics arecalculated for the states/time offsets depending on the crosscorrelations. In the specific example, the cross correlations areconverted into the logarithmic domain by applying the formulalog(0.5+p_(i)) where p_(i) is the i'th correlation value (which isbetween 0 and 1 due to the normalisation process and corresponds to aprobability that the inter time difference corresponds to the associatedtime offset).

In the example, the contribution to a given probability metric isdetermined from the previous probability metric of that time offset andthe correlation value for the offset calculated for the current frame.In addition, a contribution is made from the correlation valuesassociated with the neighbouring time offsets corresponding to thesituation where the inter time difference changes from one value toanother (i.e. such that the most probable state changes from being thatof one time offset to being that of another time offset).

The path metrics for paths from the neighbouring states corresponding toadjacent inter time difference values are weighted substantially lowerthan the path metric for the path from the same state. Specifically,experiments have shown that particular advantageous performance has beenfound for the neighbouring correlation values being weighted at leastfive times higher than the cross correlations for the same state. In thespecific example, the adjacent state path metrics are weighted by afactor of 0.009 and the same state path metric is weighted by a factorof 0.982.

FIG. 4 illustrates an example of a metric update for frame t for thetrellis state machine. In the specific example the state probabilitymetric for state s_(n) at time t is calculated from the path metric ofthe paths from the subset of previous states comprising the state s_(n)at time t−1 and the adjacent states s_(n−1) and s_(n+1) at time t−1.Specifically, the state probability metric for state s_(n) is given by:

s _(n) ^(t) =s _(n) ^(t-1) +p _(n) ^(t) +p _(n−1) ^(t) +p _(n+1) ^(t)

where p_(x) ^(t) is the calculated weighted path metric from state x tostate n in frame t.

In the example, the probability metrics are modified in each frame bysubtracting the lowest state probability metric from all stateprobability metrics. This mitigates overflow problems from continuouslyincreasing state probability metrics.

In the example, contributions to a given time offset metric are onlyincluded for the subset of offsets comprising the offset itself and theadjacent offsets. However, it will be appreciated that in otherembodiments other subsets of time offsets may be considered.

In the example, the state metrics for the trellis state machine areupdated in each frame. However, in contrast to conventional Viterbialgorithms, the state processor 207 does not select a preferred path foreach state but calculates the state probability metric for a given stateas a combined contribution from all paths entering that state. Also, thestate processor 207 does not perform a trace back through the trellis todetermine surviving paths. Rather, in the example, the current intertime difference estimate can simply be selected as the time offsetcorresponding to the state currently having the highest stateprobability metric. Thus, no delay is incurred in the state machine.Furthermore, as the probability state metric depends on previous values(and other states) a hysteris is inherently achieved.

Specifically, the state processor 207 is coupled to an ITD processor 209which determines the inter time difference from the time offsetassociated with a state having the highest state probability metric.Specifically, it may directly set the inter time difference to be equalto the time offset of the state having the highest state probabilitymetric.

The ITD processor 209 is coupled to a delay processor 211 whichdetermines the delay to be applied to the delays 109, 111. Firstly, thedelay processor 211 compensates the inter time difference by thedecimation factor applied in the decimation processor 201. In a simpleembodiment, the estimated inter time difference may be given as a numberof decimated samples (e.g. at 4 kHz corresponding to a 250 μsresolution) and this may be converted to a number of non-decimatedsamples by multiplying it by the decimation factor (e.g. to 16 kHzsamples by multiplying it by a factor of 4).

In the example, the delay processor 211 sets the values for both delays109, 111. Specifically, depending on the sign of the inter timedifference, one of the delays is set to zero and the other delay is setto the calculated number of non-decimated samples.

The described approach for calculating the inter time differenceprovides improved quality of the encoded signal and in particularprovides reduced reverberation of the mono signal prior to encoding,thereby improving the operation and performance of the CELP mono encoder115.

Specific tests have been carried out where three stereo test signalswere recorded in a conference room with a pair of microphones indifferent configurations. In the first configuration, the microphoneswere placed 1 m apart and two male talkers sat on-axis beyond each ofthe two microphones and a test conversation was recorded. In the secondconfiguration, the two microphones were placed 3 m apart and the maletalkers were again on-axis beyond each of the two microphones. In thefinal configuration, the microphones were 2 m apart and the two talkerswere broadside to the axis of the microphones but on opposite sides ofthe axis facing each of the two microphones. In all of these scenariosthe algorithm tracked the delays well and when the resultant mono signalwas encoded with the baseline algorithm for the ITU-T EV-VBR codec, again of approximately 0.3 dB in SEGSNR and WSEGSNR was observed in eachscenario.

In some embodiments, the transition from one delay to another is simplyachieved by changing the number of samples the appropriate signal isdelayed by the delays 109, 111. However, in some embodiments,functionality may be included for performing a smooth transition fromone delay to another.

Specifically, the apparatus may be arranged to transition from a firstdelay to a second delay by generating a first signal which is delayed bythe delay prior to the transition and a second signal which is delayedby the delay following the transition. The first and second signals arethen combined to generate a combined signal which includes acontribution from both the signal prior to the transition and the signalfollowing the transition. The contribution from the two signals isgradually changed such that initially the contribution is predominantlyor exclusively from the first signal and at the end of the transitionthe contribution is predominantly or exclusively from the second signal.

Thus, the apparatus may during a delay transition synthesize two signalscorresponding to the initial and the final delay. The two signals may becombined by a weighted summation such as:

s=a·s ₁ +b·s ₂

where s₁ and s₂ represent the first and second signals and a and b areweights that are modified during the transition interval (whichspecifically may be equal to a single frame). Specifically, initiallythe values may be set to a=1 and b=0 and the final values may be set toa=0 and b=1. The transition between these values may be performed inaccordance with any suitable function and may specifically maintain therelationship a+b=1 during the transition.

Thus, in such embodiments a smooth transition between different delaysis achieved by synthesizing signals for both delays and graduallytransitioning from one to the other in the time domain.

In the specific example, a 20 ms half-Hann overlap-add window is appliedto ensure that the transition from one delay to the next is asimperceptible as possible.

FIG. 5 illustrates a method of encoding a multi channel audio signal inaccordance with some embodiments of the invention.

The method initiates in step 501 wherein the multi channel audio signalcomprising at least a first audio signal from a first microphone and asecond audio signal from a second microphone is received.

Step 501 is followed by step 503 wherein an inter time differencebetween the first audio signal and the second audio signal isdetermined.

Step 503 is followed by step 505 wherein a compensated multi channelaudio signal is generated from the multi channel audio signal bydelaying at least one of the first and second stereo signals in responseto the inter time difference signal.

Step 505 is followed by step 507 wherein a mono signal is generated bycombining channels of the compensated multi channel audio signal.

Step 507 is followed by step 509 wherein the mono signal is encoded by amono signal encoder.

It will be appreciated that the above description for clarity hasdescribed embodiments of the invention with reference to differentfunctional units and processors. However, it will be apparent that anysuitable distribution of functionality between different functionalunits or processors may be used without detracting from the invention.For example, functionality illustrated to be performed by separateprocessors or controllers may be performed by the same processor orcontrollers. Hence, references to specific functional units are only tobe seen as references to suitable means for providing the describedfunctionality rather than indicative of a strict logical or physicalstructure or organization.

The invention can be implemented in any suitable form includinghardware, software, firmware or any combination of these. The inventionmay optionally be implemented at least partly as computer softwarerunning on one or more data processors and/or digital signal processors.The elements and components of an embodiment of the invention may bephysically, functionally and logically implemented in any suitable way.Indeed the functionality may be implemented in a single unit, in aplurality of units or as part of other functional units. As such, theinvention may be implemented in a single unit or may be physically andfunctionally distributed between different units and processors.

Although the present invention has been described in connection withsome embodiments, it is not intended to be limited to the specific formset forth herein. Rather, the scope of the present invention is limitedonly by the accompanying claims. Additionally, although a feature mayappear to be described in connection with particular embodiments, oneskilled in the art would recognize that various features of thedescribed embodiments may be combined in accordance with the invention.In the claims, the term comprising does not exclude the presence ofother elements or steps.

Furthermore, although individually listed, a plurality of units, means,elements or method steps may be implemented by e.g. a single unit orprocessor. Additionally, although individual features may be included indifferent claims, these may possibly be advantageously combined, and theinclusion in different claims does not imply that a combination offeatures is not feasible and/or advantageous. Also the inclusion of afeature in one category of claims does not imply a limitation to thiscategory but rather indicates that the feature is equally applicable toother claim categories as appropriate. Furthermore, the order offeatures in the claims does not imply any specific order in which thefeatures must be worked and in particular the order of individual stepsin a method claim does not imply that the steps must be performed inthis order. Rather, the steps may be performed in any suitable order.

1. An apparatus for encoding a multi channel audio signal, the apparatuscomprising: a receiver for receiving the multi channel audio signalcomprising at least a first audio signal from a first microphone and asecond audio signal from a second microphone; a time difference unit fordetermining an inter time difference between the first audio signal andthe second audio signal; a delay unit for generating a compensated multichannel audio signal from the multi channel audio signal by delaying atleast one of the first audio signal and the second audio signal inresponse to the inter time difference signal; a mono unit for generatinga mono signal by combining channels of the compensated multi channelaudio signal; and a mono signal encoder for encoding the mono signal. 2.The apparatus of claim 1 wherein the time difference unit is arranged todetermine cross correlations between the first audio signal and thesecond audio signal for a plurality of time offsets, and to determinethe inter time difference in response to the cross correlations.
 3. Theapparatus of claim 2 wherein the time difference unit is arranged to lowpass filter the first audio signal and the second audio signal prior tothe cross correlation.
 4. The apparatus of claim 2 wherein the timedifference unit is arranged to decimate the first audio signal and thesecond audio signal prior to the cross correlation.
 5. The apparatus ofclaim 2 wherein the delay unit is arranged to compensate the inter timedifference for a decimation factor of the decimation in order todetermine a delay for at least one of the first audio signal and thesecond audio signal.
 6. The apparatus of claim 2 wherein the timedifference unit is arranged to apply a spectral whitening to the firstaudio signal and the second audio signal prior to the correlation. 7.The apparatus of claim 2 wherein the time difference unit is arranged toperform windowing of the first audio signal and the second audio signalprior to the cross correlation.
 8. The apparatus of claim 2 wherein thetime difference unit comprises: a trellis state machine having aplurality of states, each of the plurality of states corresponding to atime offset of the plurality of time offsets; a path unit fordetermining path metrics for states of the trellis state machine inresponse to the cross correlations; a probability unit for determiningstate probability metrics for the states in response to path metricsassociated with paths from previous states to current states; and a unitfor determining the inter time difference in response to the stateprobability metrics.
 9. The apparatus of claim 1 wherein the delay unitis arranged to transition from a first delay to a second delay bygenerating a first compensated multi channel audio signal in response tothe first delay and a second compensated multi channel audio signal inresponse to the second delay and to combine the first compensated multichannel audio signal and second compensated multi channel audio signalto generate the compensated multi channel audio signal.
 10. A method ofencoding a multi channel audio signal, the method comprising: receivingthe multi channel audio signal comprising at least a first audio signalfrom a first microphone and a second audio signal from a secondmicrophone; determining an inter time difference between the first audiosignal and the second audio signal; generating a compensated multichannel audio signal from the multi channel audio signal by delaying atleast one of the first audio signal and the second audio signal inresponse to the inter time difference signal; generating a mono signalby combining channels of the compensated multi channel audio signal; andencoding the mono signal in a mono signal encoder.